Converged networks, in which data and voice are transmitted over the same network, are becoming increasingly popular. Transmitting voice via a data network offers several advantages over the use of a traditional voice network. For example, long-distance calls over a traditional voice network generally have a higher cost than long-distance calls over a data network. Because voice signals are transformed to data during a telephone call over a data network, calls made over a wide area data network (WAN) generally cost no more than calls made over a local area network (LAN), as the cost of sending data across a data network is generally not a function of the distance between the communicating parties.
The equipment necessary to enable voice transmission over a data network generally includes a gateway, which is a computer used to interface an analog telephone line to the data network. The gateway converts voice signals to and from the correct format for the data network protocols, and also may compress the voice data, demodulate fax signals, etc. Many different network protocols may be used to send voice data. Because of the widespread implementation of versions of the internet protocol (IP) over both LANs and WANs, IP is becoming the most common network protocol for voice transmission. The transmission of voice data over an IP network is generally referred to as VoIP.
In a VoIP telephone call, it is important to introduce as little delay as possible into the voice signal so that the telephone call does not sound choppy to the callers. While some delays are fixed, such as delays due to processing time at packet processing points and delay inherent in the physical transmission systems, other delays are variable. Sources of variable delay include queuing times at packet processing points and congestion on the network.
To achieve data transmission with as little delay as possible, current VoIP technology may employ the use of priority schemes that exist in various transport protocols, such as priority queuing at routers and bridges, to alleviate variable delay problems. However, the use of prioritization has several drawbacks. First, prioritization causes other types of data to back up at packet processing points, slowing the overall movement of data across the network. Second, prioritization routines may require the use of additional header information, increasing the size of packets and using additional network bandwidth.
Another common problem with current VoIP technology concerns the size and quantity of voice data packets generated during a typical IP telephone call. It is preferred to encapsulate short segments of voice data, typically approximately 10 milliseconds, into a single packet to preserve the natural sound of the telephone call and to avoid perceptible delays in the voice streams. However, due to efficient data compression algorithms, a packet containing this amount of data is typically much smaller than the maximum transmission unit (MTU) of a typical network component encountered by the packet as it crosses the network. Thus, a large quantity of small voice data packets may be generated per unit time during telephone calls, taking up network bandwidth with the associated large number of headers. Therefore, there remains a need for a method and a system for decreasing the amount of network resources consumed by the transmission of VoIP telephone calls.